similar to: Upgrading asterisk 13.7 to 13.11. Segfaults

06.09.2016 16:42, George Joseph ?????: > > > On Tue, Sep 6, 2016 at 7:32 AM, Dmitriy Serov <serov.d.p at gmail.com > <mailto:serov.d.p at gmail.com>> wrote: > > Hello. > > Several months server working on asterisk 13.7 and pjproject 2.5 > (installed separately). Once a day the server crashes or hangs and > is familiar sores that written

Today I faced a problem. Please help to solve this problem. Asterisk 13.7 (chan_pjsip) & Zyxel Keenetic Plus DECT, firmware v2.06(AAGJ.9)C1 Outbound call from Zyxel Keenetic (pjsip endpoint) to PSTN (pjsip trunk). Call using early media (183 Session in progress) and rtp_timeout=10. After 10 seconds: [2016-12-16 13:53:15] NOTICE[6654] res_pjsip_sdp_rtp.c: Disconnecting channel

On 20 January 2018 at 23:30, Tim S <tim.strommen at gmail.com> wrote: > I have seen this take over 2 seconds before on a sluggish machine. Thanks - my host uses SSD and everything seems pretty quick, but I'll give it a 1 second pause. > you'd need to pipe that to a Google Speech API tunnel. > That's probably not something you can hack away at with simple > Asterisk

Please help find the cause of strange behavior res_pjsip. Making outgoint call to other sip server (CommuniGatePro), my asterisk suddenly sends BYE after picking up! Partial log of an outgoing call with full debug is attached and on web: http://pastebin.com/tLNCpx4d No diagnostic messages why asterisk suddenly decided to hangup i don't found :( There are suggestions or strong belief

I think that the relevant part in C11 is section 6.2.6.1, which tells you that accessing a trap representation, _other than using a char type_, is undefined. Objects of automatic storage, which don't have an initializer are of indeterminate value, which either is an unspecified value or a trap representation. > What I found is that with -O2: > LLVM (trunk) prints both "a" and

I''m not asking for help this time! :) :) In almost every Rails project I create, I find that I want a select() popup menu that is pre-populated by data from the database. Also, I want an "Other..." option that presents a text_field_tag to input another (not presented) option (basically a combo-box). In an attempt to be as DRY as possible, I''ve come up with

Hello, I was wondering about the case below. I tried to find any information in C standard, but I found nothing. In this case, variable "i" is uninitialized, but it is the _same_ value passed as an argument, so only of "a" or "b" should be printed. What I found is that with -O2: LLVM (trunk) prints both "a" and "b" GCC (4.2) prints both

When complex assignments are performed, the R interpreter creates, then removes a special variable *tmp*. However, when byte compiling is enabled, it seems that a different mechanism for making compound assignments is used. Would it be possible to eliminate *tmp* from interpreted R code as well? It might be useful for a function to lock its own environment, and the appearance and disappearance of

Hello, I would like to add area code to local numbers, it worked like a charm on Asterisk 1.8 but does not work on Asterisk 13.11. Extensions.conf; worked before on Asterisk 1.8 ; Adding Area code to local numbers exten => _9XXXXXXX,n,Set(CALLERID(all)="$CallerID" <3818008000>) exten => _9XXXXXXX,n,Dial(SIP/voip-truck/1381${EXTEN:1},80) Any ideas? Thanks, Motty

>From this change (res_rtp_asterisk): ast 13.10 to 13.11 webrtc JSSIP stop working, failing with chan_sip.c:4083 retrans_pkt: Hanging up call 7238b48c11581d4166b899bf747a05f7 at 130.211.62.184:0 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). is there any way to configure to have the previous behaviour? Im trying to set

I have user model and referral model. Referral model has user_id as field. Now when a new user is created, I need to call referral#create as well and pass it the id of the newly generated user to user_id of referral model. How can I do that. -- You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this group, send email to

I upgraded my office installation from 13.10 to 13.11 yesterday and now I am having problems registering phones. Here is what I get on the CLI: [Sep 2 15:38:46] WARNING[2098]: res_config_mysql.c:1162 require_mysql: Realtime table general at ps_contacts: column 'qualify_timeout' cannot be type 'int(10)' (need char) [Sep 2 15:38:46] WARNING[2098]: res_config_mysql.c:1162

Can anyone give an example of the difference between the following: canreinvite=no canreinvite=yes canreinvite=update Here is the problem: I have an 800 number sent to me via SIP from a national carrier. Asterisk gets the number and rings my desk phone. Asterisk has 2 NICs, one with public IP and private IP. My phone is on private IP, the inbound call is on public. My phone rings and I answer

On 24/09/2020 12:47, Christian Naumer via samba wrote: > I am using it on Fedora with Volume Definition looking like this: and I use this: <volume fstype="cifs" ??????? server="CIFS_SERVER_FQDN" ??????? path="linprofiles" ??????? mountpoint="/mnt/%(USER)" options="username=%(USER),uid=%(USERUID),gid=%(USERGID),domain=%(DOMAIN_NAME)"

asterisk-16.13.0-rc2. Fedora 32 pjsip won't load because of undefined symbols: [Sep 4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error loading module 'func_pjsip_aor.so': /usr/lib64/asterisk/modules/func_pjsip_aor.so: undefined symbol: ast_sip_location_retrieve_aor_contacts [Sep 4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error loading module

I'm trying to install 13.3.0 from binary packages and get an error. It worked for 13.2.0 but not for 13.3.0. Centos 6.6 64bit fresh install Followed instructions on the wiki: > rpm -Uvh http://packages.asterisk.org/centos/6/current/i386/RPMS/asterisknow-version-3.0.1-2_centos6.noarch.rpm > yum update > yum install asterisk asterisk-configs --enablerepo=asterisk-13 Yum output: [goes

thank you very much. this is exactly whats needed for debug example output for your info [Dec 12 15:39:19] DEBUG[2182][C-00000000]: pjproject: <?>:         icess0x7f5d44081e88 .Added new remote candidate from the request: 2.2.2.2:57536 [Dec 12 15:39:19] DEBUG[2182][C-00000000]: pjproject: <?>:         icess0x7f5d44081e88 .New triggered check added: 1 [Dec 12 15:39:19]

On 16-08-16 17:45, George Joseph wrote: > > > On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > On 16-08-16 04:38, George Joseph wrote: >> >> >> On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens >> <jonas.kellens at telenet.be <mailto:jonas.kellens at

The current Asterisk 13 and master git branches have a new feature that will be included in 13.8.0: The ability to compile and run Asterisk with a bundled version of pjproject. ?? Why would you want to do this? Several reasons: - Predictability: When built with the ?bundled pjproject, you're always certain of the version you're running against, no matter where it's

On 06/14/2017 at 05:53 PM Joshua Colp wrote: > On Wed, Jun 14, 2017, at 12:47 PM, Michael Maier wrote: > > <snip> > >> >> I added this patch to see, if really all packages are are freed after >> they have been processed: >> >> --- b/res/res_pjsip/pjsip_distributor.c 2017-05-30 19:44:16.000000000 >> +0200 >> +++